Asterisk having trouble discerning between two different incoming lines from same SIP provider/server [SOLVED]

Asterisk Asterisk SIP
1

So I’m trying to setup two different SIP trunks going to the same server but using different usernames for different lines. One number ends in 1665 and the other ends in 1905. Unfortunately when a call comes in to my Asterisk for the 1905 number it is getting confused and using the main trunk (For the 1665 number) instead of the incoming trunk context for the 1905 number.

In short, the 1905 number should go to extension 303 (And then to the 302 Voicemail) and 1665 should go to extension 301. Instead everything just comes in on 301 no matter if either number was dialed.

Here is my sip.conf:

[general]
bindaddr=10.9.9.14
allowguest=no
disallow=all
allow=ulaw
srvlookup=yes
tcpenable=no
defaultexpiry=3600
directmedia=no
transport=udp
qualify=yes
context=internal

register => xxxxxx1665:xxx@x.x.x..82
register => xxxxxx1905:xxx@x.x.x..82

[1VoIP]
type=peer
context=from-trunk
host=x.x.x..82
defaultuser=xxxxxx1665
fromuser=xxxxxx1665
secret=xxx
insecure=port,invite
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=rfc2833
fromdomain=x.x.x..82
;qualify=yes
;maxexpiry=3600
;minexpiry=30

[1VoIP-2]
type=peer
context=from-home-trunk
host=x.x.x..82
defaultuser=xxxxxx1905
fromuser=xxxxxx1905
secret=xxx
insecure=port,invite
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=rfc2833
fromdomain=x.x.x..82

[301]
type=friend
host=dynamic
defaultuser=301
secret=xxx
context=internal
;nat=no
dtmfmode=rfc2833
canreinvite=no
callerid=301
;insecure=port,invite
mailbox=301@internal
force_rport=no
;qualify=yes

[302]
type=friend
host=dynamic
defaultuser=302
secret=xxx
context=internal-home
;nat=no
dtmfmode=rfc2833
canreinvite=no
callerid=302
;insecure=port,invite
mailbox=302@internal
;qualify=yes

[303]
type=friend
host=dynamic
defaultuser=303
secret=xxx
context=internal-home
;nat=no
dtmfmode=rfc2833
canreinvite=no
callerid=303
;insecure=port,invite
mailbox=303@internal

extensions.conf

[general]
static=yes
writeprotect=no
clearglobalvars=no

[from-trunk]
exten => s,1,GotoIf(${DB_EXISTS(blacklist/${CALLERID(num)})}?blacklisted,s,1)
exten => s,2,Dial(SIP/301,20)
exten => s,3,Dial(SIP/1VoIP/xxxxxxxxxx,20)
exten => s,4,VoiceMail(301@internal,u)
exten => s,5,Hangup()

[from-home-trunk]
exten => s,1,GotoIf(${DB_EXISTS(blacklist/${CALLERID(num)})}?blacklisted,s,1)
exten => s,2,Dial(SIP/303,40)
exten => s,3,VoiceMail(302@internal,u)
exten => s,4,Hangup()

[blacklisted]
exten => s,1,Answer()
exten => s,2,Wait(3)
exten => s,3,Playback(tt-allbusy)
exten => s,4,Wait(10)
exten => s,5,Playback(tt-allbusy)
exten => s,6,Wait(10)
exten => s,7,Playback(tt-somethingwrong)
exten => s,8,Playback(abandon-all-hope)
exten => s,9,Wait(10)
exten => s,10,Playback(tt-weasels)
exten => s,11,Wait(5)
exten => s,12,Playback(tt-allbusy)
exten => s,13,Wait(5)
exten => s,14,Playback(tt-monkeysintro)
exten => s,15,Playback(tt-monkeys)
exten => s,16,Wait(10)
exten => s,17,Playback(tt-monty-knights)
exten => s,18,Playback(hal_goodbye)
exten => s,19,Hangup()

[internal]
exten => _XXXXXXXXXX,1,Dial(SIP/${EXTEN}@1VoIP)
exten => _XXXXXXXXXXX,1,Dial(SIP/${EXTEN}@1VoIP)
exten => 399,1,VoiceMailMain(${CALLERID(num)}@internal,u)
exten => 399,2,Hangup

[internal-home]
exten => _XXXXXXXXXX,1,Dial(SIP/${EXTEN}@1VoIP-2)
exten => _XXXXXXXXXXX,1,Dial(SIP/${EXTEN}@1VoIP-2)
exten => 399,1,VoiceMailMain(${CALLERID(num)}@internal,u)
exten => 399,2,Hangup
2

<— SIP read from UDP:x.x.x.82:5060 —>
INVITE sip:s@10.9.9.14:5060 SIP/2.0
Record-Route: sip:x.x.x.82;lr=on;ftag=as5b94a9c1;nat=yes
Via: SIP/2.0/UDP x.x.x.82;branch=z9hG4bK6053.ecdcc372.0
Via: SIP/2.0/UDP x.x.x.38:5060;branch=z9hG4bK56d9a1f5;rport=5060
Max-Forwards: 69
From: “EULESS TX” sip:xxxxxx0289@x.x.x.38;tag=as5b94a9c1
To: sip:xxxxxx1905@x.x.x.82:5060
Contact: sip:xxxxxx0289@x.x.x.38:5060
Call-ID: 20357f0819f91c9b142da21125146c08@x.x.x.38:5060
CSeq: 102 INVITE
User-Agent: VYLmedia/HostedPBX-2.0.0
Date: Wed, 22 Nov 2017 21:35:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: “EULESS TX” sip:xxxxxx0289@x.x.x.38;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 280

v=0
o=root 936390204 936390204 IN IP4 x.x.x.82
s=Asterisk PBX 11.22.0
c=IN IP4 x.x.x.82
t=0 0
m=audio 57652 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=nortpproxy:yes
<------------->
— (17 headers 13 lines) —
Sending to x.x.x.82:5060 (no NAT)
Sending to x.x.x.82:5060 (no NAT)
Using INVITE request as basis request - 20357f0819f91c9b142da21125146c08@x.x.x.38:5060
Found peer ‘1VoIP’ for ‘xxxxxx0289’ from x.x.x.82:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0x7f6c3401c7f0 – Strict RTP learning after remote address set to: x.x.x.82:57652
Peer audio RTP is at port x.x.x.82:57652
Looking for s in from-trunk (domain 10.9.9.14)
sip_route_dump: route/path hop: sip:x.x.x.82;lr=on;ftag=as5b94a9c1;nat=yes

<— Transmitting (no NAT) to x.x.x.82:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP x.x.x.82;branch=z9hG4bK6053.ecdcc372.0;received=x.x.x.82
Via: SIP/2.0/UDP x.x.x.38:5060;branch=z9hG4bK56d9a1f5;rport=5060
Record-Route: sip:x.x.x.82;lr=on;ftag=as5b94a9c1;nat=yes
From: “EULESS TX” sip:xxxxxx0289@x.x.x.38;tag=as5b94a9c1
To: sip:xxxxxx1905@x.x.x.82:5060
Call-ID: 20357f0819f91c9b142da21125146c08@x.x.x.38:5060
CSeq: 102 INVITE
Server: Asterisk PBX 13.18.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:s@10.9.9.14:5060
Content-Length: 0

<------------>
– Executing [s@from-trunk:1] GotoIf(“SIP/1VoIP-00000000”, “0?blacklisted,s,1”) in new stack
– Executing [s@from-trunk:2] Dial(“SIP/1VoIP-00000000”, “SIP/301,20”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 10118
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.9.9.4:5060:
INVITE sip:301@10.9.9.4:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK356dbc76
Max-Forwards: 70
From: “EULESS TX” sip:xxxxxx0289@10.9.9.14;tag=as4b1f7b07
To: sip:301@10.9.9.4:5060;transport=udp
Contact: sip:xxxxxx0289@10.9.9.14:5060
Call-ID: 433c965b39e780fb237e6aab61c6c75b@10.9.9.14:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.18.2
Date: Wed, 22 Nov 2017 21:35:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 246

v=0
o=root 1025142561 1025142561 IN IP4 10.9.9.14
s=Asterisk PBX 13.18.2
c=IN IP4 10.9.9.14
t=0 0
m=audio 10118 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

-- Called SIP/301

<— SIP read from UDP:10.9.9.4:51724 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK356dbc76
From: “EULESS TX” sip:xxxxxx0289@10.9.9.14;tag=as4b1f7b07
To: sip:301@10.9.9.4:5060;transport=udp
Call-ID: 433c965b39e780fb237e6aab61c6c75b@10.9.9.14:5060
Date: Wed, 22 Nov 2017 21:35:55 GMT
CSeq: 102 INVITE
Server: Cisco-CP7961G/8.5.3
Contact: sip:301@10.9.9.4:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Allow-Events: kpml,dialog
Content-Length: 0

<------------->
— (12 headers 0 lines) —

<— SIP read from UDP:10.9.9.4:51725 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK356dbc76
From: “EULESS TX” sip:xxxxxx0289@10.9.9.14;tag=as4b1f7b07
To: sip:301@10.9.9.4:5060;transport=udp;tag=00229005b18d09740629d0d3-c9d96348
Call-ID: 433c965b39e780fb237e6aab61c6c75b@10.9.9.14:5060
Date: Wed, 22 Nov 2017 21:35:55 GMT
CSeq: 102 INVITE
Server: Cisco-CP7961G/8.5.3
Contact: sip:301@10.9.9.4:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Remote-Party-ID: “301” sip:301@10.9.9.14;party=called;id-type=subscriber;privacy=off;screen=yes
Allow-Events: kpml,dialog
Content-Length: 0

<------------->
— (13 headers 0 lines) —
sip_route_dump: route/path hop: sip:301@10.9.9.4:5060;transport=udp
– SIP/301-00000001 is ringing

<— Transmitting (no NAT) to x.x.x.82:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP x.x.x.82;branch=z9hG4bK6053.ecdcc372.0;received=x.x.x.82
Via: SIP/2.0/UDP x.x.x.38:5060;branch=z9hG4bK56d9a1f5;rport=5060
Record-Route: sip:x.x.x.82;lr=on;ftag=as5b94a9c1;nat=yes
From: “EULESS TX” sip:xxxxxx0289@x.x.x.38;tag=as5b94a9c1
To: sip:xxxxxx1905@x.x.x.82:5060;tag=as27ca9925
Call-ID: 20357f0819f91c9b142da21125146c08@x.x.x.38:5060
CSeq: 102 INVITE
Server: Asterisk PBX 13.18.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:s@10.9.9.14:5060
Content-Length: 0

<------------>

<— SIP read from UDP:x.x.x.82:5060 —>
CANCEL sip:s@10.9.9.14:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.82;branch=z9hG4bK6053.ecdcc372.0
Max-Forwards: 69
From: “EULESS TX” sip:xxxxxx0289@x.x.x.38;tag=as5b94a9c1
To: sip:xxxxxx1905@x.x.x.82:5060
Call-ID: 20357f0819f91c9b142da21125146c08@x.x.x.38:5060
CSeq: 102 CANCEL
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Sending to x.x.x.82:5060 (no NAT)

<— Reliably Transmitting (no NAT) to x.x.x.82:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP x.x.x.82;branch=z9hG4bK6053.ecdcc372.0;received=x.x.x.82
Via: SIP/2.0/UDP x.x.x.38:5060;branch=z9hG4bK56d9a1f5;rport=5060
From: “EULESS TX” sip:xxxxxx0289@x.x.x.38;tag=as5b94a9c1
To: sip:xxxxxx1905@x.x.x.82:5060;tag=as27ca9925
Call-ID: 20357f0819f91c9b142da21125146c08@x.x.x.38:5060
CSeq: 102 INVITE
Server: Asterisk PBX 13.18.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>

<— Transmitting (no NAT) to x.x.x.82:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.x.x.82;branch=z9hG4bK6053.ecdcc372.0;received=x.x.x.82
From: “EULESS TX” sip:xxxxxx0289@x.x.x.38;tag=as5b94a9c1
To: sip:xxxxxx1905@x.x.x.82:5060;tag=as27ca9925
Call-ID: 20357f0819f91c9b142da21125146c08@x.x.x.38:5060
CSeq: 102 CANCEL
Server: Asterisk PBX 13.18.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘433c965b39e780fb237e6aab61c6c75b@10.9.9.14:5060’ in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 10.9.9.4:5060:
CANCEL sip:301@10.9.9.4:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK356dbc76
Max-Forwards: 70
From: “EULESS TX” sip:xxxxxx0289@10.9.9.14;tag=as4b1f7b07
To: sip:301@10.9.9.4:5060;transport=udp
Call-ID: 433c965b39e780fb237e6aab61c6c75b@10.9.9.14:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 13.18.2
Content-Length: 0

Scheduling destruction of SIP dialog ‘433c965b39e780fb237e6aab61c6c75b@10.9.9.14:5060’ in 6400 ms (Method: INVITE)
== Spawn extension (from-trunk, s, 2) exited non-zero on ‘SIP/1VoIP-00000000’

<— SIP read from UDP:x.x.x.82:5060 —>
ACK sip:s@10.9.9.14:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.82;branch=z9hG4bK6053.ecdcc372.0
Max-Forwards: 69
From: “EULESS TX” sip:xxxxxx0289@x.x.x.38;tag=as5b94a9c1
To: sip:xxxxxx1905@x.x.x.82:5060;tag=as27ca9925
Call-ID: 20357f0819f91c9b142da21125146c08@x.x.x.38:5060
CSeq: 102 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘20357f0819f91c9b142da21125146c08@x.x.x.38:5060’ Method: ACK

<— SIP read from UDP:10.9.9.4:51726 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK356dbc76
From: “EULESS TX” sip:xxxxxx0289@10.9.9.14;tag=as4b1f7b07
To: sip:301@10.9.9.4:5060;transport=udp;tag=00229005b18d09740629d0d3-c9d96348
Call-ID: 433c965b39e780fb237e6aab61c6c75b@10.9.9.14:5060
Date: Wed, 22 Nov 2017 21:36:01 GMT
CSeq: 102 CANCEL
Server: Cisco-CP7961G/8.5.3
Content-Length: 0

<------------->
— (9 headers 0 lines) —

<— SIP read from UDP:10.9.9.4:51727 —>
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK356dbc76
From: “EULESS TX” sip:xxxxxx0289@10.9.9.14;tag=as4b1f7b07
To: sip:301@10.9.9.4:5060;transport=udp;tag=00229005b18d09740629d0d3-c9d96348
Call-ID: 433c965b39e780fb237e6aab61c6c75b@10.9.9.14:5060
Date: Wed, 22 Nov 2017 21:36:01 GMT
CSeq: 102 INVITE
Server: Cisco-CP7961G/8.5.3
Contact: sip:301@10.9.9.4:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Remote-Party-ID: “301” sip:301@10.9.9.14;party=called;id-type=subscriber;privacy=off;screen=yes
Allow-Events: kpml,dialog
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Transmitting (no NAT) to 10.9.9.4:5060:
ACK sip:301@10.9.9.4:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK356dbc76
Max-Forwards: 70
From: “EULESS TX” sip:xxxxxx0289@10.9.9.14;tag=as4b1f7b07
To: sip:301@10.9.9.4:5060;transport=udp;tag=00229005b18d09740629d0d3-c9d96348
Contact: sip:xxxxxx0289@10.9.9.14:5060
Call-ID: 433c965b39e780fb237e6aab61c6c75b@10.9.9.14:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.18.2
Content-Length: 0

Scheduling destruction of SIP dialog ‘433c965b39e780fb237e6aab61c6c75b@10.9.9.14:5060’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:x.x.x.82:5060 —>

<------------->
Really destroying SIP dialog ‘433c965b39e780fb237e6aab61c6c75b@10.9.9.14:5060’ Method: INVITE
Really destroying SIP dialog ‘2ab223b1751d8281751bd2ac7b0b8c42@x.x.x.38:5060’ Method: NOTIFY
Really destroying SIP dialog ‘40d7a8ac514448f7754520e76a342942@208.80.13.37:5060’ Method: NOTIFY
Really destroying SIP dialog ‘42c6bb38763634230c8f861c50516654@x.x.x.37:5060’ Method: NOTIFY
Really destroying SIP dialog ‘5ddc4f6529c1001a0417f2564f450a4e@208.80.13.38:5060’ Method: NOTIFY
Really destroying SIP dialog ‘4485012c616d47bc0a6ca28324d8cbb3@x.x.x.37:5060’ Method: NOTIFY
Really destroying SIP dialog ‘645be3ae6007b8cf2fb56fab4d695e56@x.x.x.38:5060’ Method: NOTIFY
Really destroying SIP dialog ‘6619a0bc73d209925bf8d4497608e157@208.80.13.37:5060’ Method: NOTIFY
Really destroying SIP dialog ‘4182eb2447daea6d1cc20de86fb32bfe@208.80.13.38:5060’ Method: NOTIFY

<— SIP read from UDP:x.x.x.82:5060 —>

<------------->
Reliably Transmitting (no NAT) to x.x.x.82:5060:
OPTIONS sip:x.x.x.82 SIP/2.0
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK79ab863f
Max-Forwards: 70
From: “asterisk” sip:xxxxxx1905@10.9.9.14;tag=as3a7e996a
To: sip:x.x.x.82
Contact: sip:xxxxxx1905@10.9.9.14:5060
Call-ID: 435d88480656632e5ebe1f3c3635bd99@10.9.9.14:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.18.2
Date: Wed, 22 Nov 2017 21:36:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:x.x.x.82:5060 —>
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK79ab863f;rport=5060;received=x.x.x.233
From: “asterisk” sip:xxxxxx1905@10.9.9.14;tag=as3a7e996a
To: sip:x.x.x.82;tag=b96d0dd7209d7834958cacf8eb0d0ba5.f08f
Call-ID: 435d88480656632e5ebe1f3c3635bd99@10.9.9.14:5060
CSeq: 102 OPTIONS
Server: kamailio (3.3.1 (x86_64/linux))
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘435d88480656632e5ebe1f3c3635bd99@10.9.9.14:5060’ Method: OPTIONS
Reliably Transmitting (no NAT) to x.x.x.82:5060:
OPTIONS sip:x.x.x.82 SIP/2.0
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK22f2a20e
Max-Forwards: 70
From: “asterisk” sip:xxxxxx1665@10.9.9.14;tag=as5bb808cb
To: sip:x.x.x.82
Contact: sip:xxxxxx1665@10.9.9.14:5060
Call-ID: 0adf201d290997d27ff2fb455a636bca@10.9.9.14:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.18.2
Date: Wed, 22 Nov 2017 21:36:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:x.x.x.82:5060 —>
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK22f2a20e;rport=5060;received=x.x.x.233
From: “asterisk” sip:xxxxxx1665@10.9.9.14;tag=as5bb808cb
To: sip:x.x.x.82;tag=b96d0dd7209d7834958cacf8eb0d0ba5.8b72
Call-ID: 0adf201d290997d27ff2fb455a636bca@10.9.9.14:5060
CSeq: 102 OPTIONS
Server: kamailio (3.3.1 (x86_64/linux))
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘0adf201d290997d27ff2fb455a636bca@10.9.9.14:5060’ Method: OPTIONS
Reliably Transmitting (no NAT) to 10.9.9.4:5060:
OPTIONS sip:302@10.9.9.4:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK383001ce
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.9.9.14;tag=as7cf5f07b
To: sip:302@10.9.9.4:5060;transport=udp
Contact: sip:asterisk@10.9.9.14:5060
Call-ID: 5e06955e475ba1af1f39af505fe0b76d@10.9.9.14:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.18.2
Date: Wed, 22 Nov 2017 21:36:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:10.9.9.4:51728 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK383001ce
From: “asterisk” sip:asterisk@10.9.9.14;tag=as7cf5f07b
To: sip:302@10.9.9.4:5060;transport=udp;tag=00229005b18d0975621446cf-addb3f14
Call-ID: 5e06955e475ba1af1f39af505fe0b76d@10.9.9.14:5060
Date: Wed, 22 Nov 2017 21:36:37 GMT
CSeq: 102 OPTIONS
Server: Cisco-CP7961G/8.5.3
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Allow-Events: kpml,dialog,refer
Accept: application/sdp,multipart/mixed,multipart/alternative
Accept-Encoding: identity
Accept-Language: en
Supported: replaces,join,norefersub
Content-Length: 233
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 26388 0 IN IP4 10.9.9.4
s=SIP Call
t=0 0
m=audio 0 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (17 headers 11 lines) —
Really destroying SIP dialog ‘5e06955e475ba1af1f39af505fe0b76d@10.9.9.14:5060’ Method: OPTIONS
Reliably Transmitting (no NAT) to 10.9.9.2:5060:
OPTIONS sip:303@10.9.9.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK30c8797f
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.9.9.14;tag=as45f87bca
To: sip:303@10.9.9.2:5060
Contact: sip:asterisk@10.9.9.14:5060
Call-ID: 266d1b8f0efdf2be056476893e6d39c1@10.9.9.14:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.18.2
Date: Wed, 22 Nov 2017 21:36:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Reliably Transmitting (no NAT) to 10.9.9.4:5060:
OPTIONS sip:301@10.9.9.4:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK504723c5
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.9.9.14;tag=as45217a74
To: sip:301@10.9.9.4:5060;transport=udp
Contact: sip:asterisk@10.9.9.14:5060
Call-ID: 1892d2d439b52db923d65d9837cd70a3@10.9.9.14:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.18.2
Date: Wed, 22 Nov 2017 21:36:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:10.9.9.2:5060 —>
SIP/2.0 200 OK
To: sip:303@10.9.9.2:5060;tag=5580ed847c2b8b47i0
From: “asterisk” sip:asterisk@10.9.9.14;tag=as45f87bca
Call-ID: 266d1b8f0efdf2be056476893e6d39c1@10.9.9.14:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK30c8797f
Server: Cisco/SPA112-1.4.1SR1(002)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘266d1b8f0efdf2be056476893e6d39c1@10.9.9.14:5060’ Method: OPTIONS

<— SIP read from UDP:10.9.9.4:51729 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK504723c5
From: “asterisk” sip:asterisk@10.9.9.14;tag=as45217a74
To: sip:301@10.9.9.4:5060;transport=udp;tag=00229005b18d09765a9142c0-d704b5bd
Call-ID: 1892d2d439b52db923d65d9837cd70a3@10.9.9.14:5060
Date: Wed, 22 Nov 2017 21:36:37 GMT
CSeq: 102 OPTIONS
Server: Cisco-CP7961G/8.5.3
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Allow-Events: kpml,dialog,refer
Accept: application/sdp,multipart/mixed,multipart/alternative
Accept-Encoding: identity
Accept-Language: en
Supported: replaces,join,norefersub
Content-Length: 233
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 13694 0 IN IP4 10.9.9.4
s=SIP Call
t=0 0
m=audio 0 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (17 headers 11 lines) —
Really destroying SIP dialog ‘1892d2d439b52db923d65d9837cd70a3@10.9.9.14:5060’ Method: OPTIONS

<— SIP read from UDP:x.x.x.82:5060 —>

<------------->

<— SIP read from UDP:x.x.x.82:5060 —>

<------------->
Reliably Transmitting (no NAT) to x.x.x.82:5060:
OPTIONS sip:x.x.x.82 SIP/2.0
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK3cd00737
Max-Forwards: 70
From: “asterisk” sip:xxxxxx1905@10.9.9.14;tag=as06dac5c1
To: sip:x.x.x.82
Contact: sip:xxxxxx1905@10.9.9.14:5060
Call-ID: 763e0dc30257f12b7bef86370b94b0a6@10.9.9.14:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.18.2
Date: Wed, 22 Nov 2017 21:37:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:x.x.x.82:5060 —>
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK3cd00737;rport=5060;received=x.x.x.233
From: “asterisk” sip:xxxxxx1905@10.9.9.14;tag=as06dac5c1
To: sip:x.x.x.82;tag=b96d0dd7209d7834958cacf8eb0d0ba5.c2ef
Call-ID: 763e0dc30257f12b7bef86370b94b0a6@10.9.9.14:5060
CSeq: 102 OPTIONS
Server: kamailio (3.3.1 (x86_64/linux))
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘763e0dc30257f12b7bef86370b94b0a6@10.9.9.14:5060’ Method: OPTIONS
Reliably Transmitting (no NAT) to x.x.x.82:5060:
OPTIONS sip:x.x.x.82 SIP/2.0
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK39988d8f
Max-Forwards: 70
From: “asterisk” sip:xxxxxx1665@10.9.9.14;tag=as492a6d35
To: sip:x.x.x.82
Contact: sip:xxxxxx1665@10.9.9.14:5060
Call-ID: 25f4ecad1b9167cb00b6be4e51bfe339@10.9.9.14:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.18.2
Date: Wed, 22 Nov 2017 21:37:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:x.x.x.82:5060 —>
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK39988d8f;rport=5060;received=x.x.x.233
From: “asterisk” sip:xxxxxx1665@10.9.9.14;tag=as492a6d35
To: sip:x.x.x.82;tag=b96d0dd7209d7834958cacf8eb0d0ba5.1d46
Call-ID: 25f4ecad1b9167cb00b6be4e51bfe339@10.9.9.14:5060
CSeq: 102 OPTIONS
Server: kamailio (3.3.1 (x86_64/linux))
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘25f4ecad1b9167cb00b6be4e51bfe339@10.9.9.14:5060’ Method: OPTIONS
Reliably Transmitting (no NAT) to 10.9.9.4:5060:
OPTIONS sip:302@10.9.9.4:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK3370c16e
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.9.9.14;tag=as379e3c68
To: sip:302@10.9.9.4:5060;transport=udp
Contact: sip:asterisk@10.9.9.14:5060
Call-ID: 0db4b32e15230de94195e79d286218e0@10.9.9.14:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.18.2
Date: Wed, 22 Nov 2017 21:37:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:10.9.9.4:51730 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK3370c16e
From: “asterisk” sip:asterisk@10.9.9.14;tag=as379e3c68
To: sip:302@10.9.9.4:5060;transport=udp;tag=00229005b18d0977619fb890-f209d060
Call-ID: 0db4b32e15230de94195e79d286218e0@10.9.9.14:5060
Date: Wed, 22 Nov 2017 21:37:37 GMT
CSeq: 102 OPTIONS
Server: Cisco-CP7961G/8.5.3
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Allow-Events: kpml,dialog,refer
Accept: application/sdp,multipart/mixed,multipart/alternative
Accept-Encoding: identity
Accept-Language: en
Supported: replaces,join,norefersub
Content-Length: 232
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 1938 0 IN IP4 10.9.9.4
s=SIP Call
t=0 0
m=audio 0 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (17 headers 11 lines) —
Really destroying SIP dialog ‘0db4b32e15230de94195e79d286218e0@10.9.9.14:5060’ Method: OPTIONS
Reliably Transmitting (no NAT) to 10.9.9.2:5060:
OPTIONS sip:303@10.9.9.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK51250dba
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.9.9.14;tag=as77bea662
To: sip:303@10.9.9.2:5060
Contact: sip:asterisk@10.9.9.14:5060
Call-ID: 73c39c4d39554ff211ba114a12af9e8a@10.9.9.14:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.18.2
Date: Wed, 22 Nov 2017 21:37:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Reliably Transmitting (no NAT) to 10.9.9.4:5060:
OPTIONS sip:301@10.9.9.4:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK1eb0a770
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.9.9.14;tag=as77c3d594
To: sip:301@10.9.9.4:5060;transport=udp
Contact: sip:asterisk@10.9.9.14:5060
Call-ID: 184be74661a03fc372d865e73e8f8dc8@10.9.9.14:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.18.2
Date: Wed, 22 Nov 2017 21:37:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:10.9.9.4:51731 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK1eb0a770
From: “asterisk” sip:asterisk@10.9.9.14;tag=as77c3d594
To: sip:301@10.9.9.4:5060;transport=udp;tag=00229005b18d0978ecce6da0-bb57c5b0
Call-ID: 184be74661a03fc372d865e73e8f8dc8@10.9.9.14:5060
Date: Wed, 22 Nov 2017 21:37:37 GMT
CSeq: 102 OPTIONS
Server: Cisco-CP7961G/8.5.3
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Allow-Events: kpml,dialog,refer
Accept: application/sdp,multipart/mixed,multipart/alternative
Accept-Encoding: identity
Accept-Language: en
Supported: replaces,join,norefersub
Content-Length: 232
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 2762 0 IN IP4 10.9.9.4
s=SIP Call
t=0 0
m=audio 0 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (17 headers 11 lines) —
Really destroying SIP dialog ‘184be74661a03fc372d865e73e8f8dc8@10.9.9.14:5060’ Method: OPTIONS

<— SIP read from UDP:10.9.9.2:5060 —>
SIP/2.0 200 OK
To: sip:303@10.9.9.2:5060;tag=5580ed847c2b8b47i0
From: “asterisk” sip:asterisk@10.9.9.14;tag=as77bea662
Call-ID: 73c39c4d39554ff211ba114a12af9e8a@10.9.9.14:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK51250dba
Server: Cisco/SPA112-1.4.1SR1(002)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘73c39c4d39554ff211ba114a12af9e8a@10.9.9.14:5060’ Method: OPTIONS

<— SIP read from UDP:x.x.x.82:5060 —>

<------------->

<— SIP read from UDP:x.x.x.82:5060 —>

<------------->
Reliably Transmitting (no NAT) to x.x.x.82:5060:
OPTIONS sip:x.x.x.82 SIP/2.0
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK71d6356a
Max-Forwards: 70
From: “asterisk” sip:xxxxxx1905@10.9.9.14;tag=as2c691515
To: sip:x.x.x.82
Contact: sip:xxxxxx1905@10.9.9.14:5060
Call-ID: 0be70a6f377daa7d31399c496383fae9@10.9.9.14:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.18.2
Date: Wed, 22 Nov 2017 21:38:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:x.x.x.82:5060 —>
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK71d6356a;rport=5060;received=x.x.x.233
From: “asterisk” sip:xxxxxx1905@10.9.9.14;tag=as2c691515
To: sip:x.x.x.82;tag=b96d0dd7209d7834958cacf8eb0d0ba5.06f4
Call-ID: 0be70a6f377daa7d31399c496383fae9@10.9.9.14:5060
CSeq: 102 OPTIONS
Server: kamailio (3.3.1 (x86_64/linux))
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘0be70a6f377daa7d31399c496383fae9@10.9.9.14:5060’ Method: OPTIONS
Reliably Transmitting (no NAT) to x.x.x.82:5060:
OPTIONS sip:x.x.x.82 SIP/2.0
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK4cc31283
Max-Forwards: 70
From: “asterisk” sip:xxxxxx1665@10.9.9.14;tag=as00d1758f
To: sip:x.x.x.82
Contact: sip:xxxxxx1665@10.9.9.14:5060
Call-ID: 48af56472f44d8072be08e83205c038a@10.9.9.14:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.18.2
Date: Wed, 22 Nov 2017 21:38:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:x.x.x.82:5060 —>
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK4cc31283;rport=5060;received=x.x.x.233
From: “asterisk” sip:xxxxxx1665@10.9.9.14;tag=as00d1758f
To: sip:x.x.x.82;tag=b96d0dd7209d7834958cacf8eb0d0ba5.5b4e
Call-ID: 48af56472f44d8072be08e83205c038a@10.9.9.14:5060
CSeq: 102 OPTIONS
Server: kamailio (3.3.1 (x86_64/linux))
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘48af56472f44d8072be08e83205c038a@10.9.9.14:5060’ Method: OPTIONS
Reliably Transmitting (no NAT) to 10.9.9.4:5060:
OPTIONS sip:302@10.9.9.4:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK504c8c3f
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.9.9.14;tag=as4fdb9c01
To: sip:302@10.9.9.4:5060;transport=udp
Contact: sip:asterisk@10.9.9.14:5060
Call-ID: 4a43d666662c23b25c27fce30e9f4d3a@10.9.9.14:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.18.2
Date: Wed, 22 Nov 2017 21:38:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:10.9.9.4:51732 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK504c8c3f
From: “asterisk” sip:asterisk@10.9.9.14;tag=as4fdb9c01
To: sip:302@10.9.9.4:5060;transport=udp;tag=00229005b18d0979bdd346bb-e741c586
Call-ID: 4a43d666662c23b25c27fce30e9f4d3a@10.9.9.14:5060
Date: Wed, 22 Nov 2017 21:38:37 GMT
CSeq: 102 OPTIONS
Server: Cisco-CP7961G/8.5.3
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Allow-Events: kpml,dialog,refer
Accept: application/sdp,multipart/mixed,multipart/alternative
Accept-Encoding: identity
Accept-Language: en
Supported: replaces,join,norefersub
Content-Length: 233
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 14777 0 IN IP4 10.9.9.4
s=SIP Call
t=0 0
m=audio 0 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (17 headers 11 lines) —
Really destroying SIP dialog ‘4a43d666662c23b25c27fce30e9f4d3a@10.9.9.14:5060’ Method: OPTIONS
Reliably Transmitting (no NAT) to 10.9.9.4:5060:
OPTIONS sip:301@10.9.9.4:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK0ebbffdc
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.9.9.14;tag=as0e7a6ab4
To: sip:301@10.9.9.4:5060;transport=udp
Contact: sip:asterisk@10.9.9.14:5060
Call-ID: 1d68cbda1fa7334126a0004909f46b13@10.9.9.14:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.18.2
Date: Wed, 22 Nov 2017 21:38:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Reliably Transmitting (no NAT) to 10.9.9.2:5060:
OPTIONS sip:303@10.9.9.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK656f16cd
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.9.9.14;tag=as58ebfa43
To: sip:303@10.9.9.2:5060
Contact: sip:asterisk@10.9.9.14:5060
Call-ID: 10a737511a15e13b3935785f6dcc212a@10.9.9.14:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.18.2
Date: Wed, 22 Nov 2017 21:38:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:10.9.9.4:51733 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK0ebbffdc
From: “asterisk” sip:asterisk@10.9.9.14;tag=as0e7a6ab4
To: sip:301@10.9.9.4:5060;transport=udp;tag=00229005b18d097a921740aa-624e3271
Call-ID: 1d68cbda1fa7334126a0004909f46b13@10.9.9.14:5060
Date: Wed, 22 Nov 2017 21:38:37 GMT
CSeq: 102 OPTIONS
Server: Cisco-CP7961G/8.5.3
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Allow-Events: kpml,dialog,refer
Accept: application/sdp,multipart/mixed,multipart/alternative
Accept-Encoding: identity
Accept-Language: en
Supported: replaces,join,norefersub
Content-Length: 233
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 16095 0 IN IP4 10.9.9.4
s=SIP Call
t=0 0
m=audio 0 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (17 headers 11 lines) —
Really destroying SIP dialog ‘1d68cbda1fa7334126a0004909f46b13@10.9.9.14:5060’ Method: OPTIONS

<— SIP read from UDP:10.9.9.2:5060 —>
SIP/2.0 200 OK
To: sip:303@10.9.9.2:5060;tag=5580ed847c2b8b47i0
From: “asterisk” sip:asterisk@10.9.9.14;tag=as58ebfa43
Call-ID: 10a737511a15e13b3935785f6dcc212a@10.9.9.14:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.9.9.14:5060;branch=z9hG4bK656f16cd
Server: Cisco/SPA112-1.4.1SR1(002)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘10a737511a15e13b3935785f6dcc212a@10.9.9.14:5060’ Method: OPTIONS

<— SIP read from UDP:x.x.x.82:5060 —>

<------------->

3

Type=peer matches on IP address. It might also match on port, but you have insecure=port (and, in this case, need it).

Consequently the behaviour is expected.

If you want to discriminate, you must use different callback extensions (i.e. the request URI user must be present and differ), assuming that the peer supports that on registration. The only other automatic way is if they can be requested to put the account name in the from user, in which case you can use type=friend, but note that the insecure=invite opens a means of attacking your system. They to have RPID, so you won’t lose caller ID, although you’ll need to enable that.

Finally, if the To header contains the account information, you can manually extract that, once the dialplan has started, and route on that.

canreinvite is an obsolete synonym for directmedia, so it makes no sense to have both in your configuration.

1 Like
4

Thanks for the canreinvite tip david.

Hmm it looks like parsing the To header is probably going to be my best bet, or getting them to put my 2nd number on a different server…

Would something like this work?

[from-trunk]
exten => s,1,GotoIf(${DB_EXISTS(blacklist/${CALLERID(num)})}?blacklisted,s,1)
exten => s,2,Set(thedid=${SIP_HEADER(To)})
exten => s,3,Set(thedid=${CUT(thedid,@,1)})
exten => s,4,Set(thedid=${CUT(thedid,:,2)})
exten => s,5,GotoIf($["${thedid}" = "xxxxxx1665"]?from-work-trunk,s,1)
exten => s,6,GotoIf($["${thedid}" = "xxxxxx1905"]?from-home-trunk,s,1)

[from-work-trunk]
exten => s,1,Dial(SIP/301,20)
exten => s,2,Dial(SIP/1VoIP/xxxxxx0289,20)
exten => s,3,VoiceMail(301@internal,u)
exten => s,4,Hangup()

[from-home-trunk]
exten => s,1,Dial(SIP/303,40)
exten => s,2,VoiceMail(302@internal,u)
exten => s,3,Hangup()
5

Tested and seems to work fine.